Team,
We are deployed Wazo-platform on AWS Ec2 instance with Debian 10 buster.
Instance type m5.large.
Issue : When we try to make a call from Wazo WebRTC sdk, call was connected but audio not working.
Can you help me to resolve this issues.
Thanks in advance.
To help you we need pcap or logs from the call. Check your SDP.
We are also facing same issue,
i have attached the pjsip history log for reference
there is no rtp packet flow in that call.
== Manager ‘xivo_munin_user’ logged off from 127.0.0.1
ip-172-31-56-122CLI> pjsip show history
No. Timestamp (Dir) Address SIP Message
===== ========== ============================== ===================================
00000 1626865222 * ==> 127.0.0.1:58130 OPTIONS sip:utq139nj@127.0.0.1:58130;transport=ws SIP/2.0
00001 1626865223 * <== 127.0.0.1:58130 SIP/2.0 200 OK
00002 1626865230 * <== 127.0.0.1:33662 REGISTER sip:ucpm.s.com SIP/2.0
00003 1626865230 * ==> 127.0.0.1:33662 SIP/2.0 401 Unauthorized
00004 1626865230 * ==> 127.0.0.1:59694 OPTIONS sip:jb98sgm8@127.0.0.1:59694;transport=ws SIP/2.0
00005 1626865230 * <== 127.0.0.1:59694 SIP/2.0 200 OK
00006 1626865231 * <== 127.0.0.1:33662 REGISTER sip:ucpm.s.com SIP/2.0
00007 1626865231 * ==> 127.0.0.1:33662 SIP/2.0 200 OK
00008 1626865231 * ==> 127.0.0.1:48342 OPTIONS sip:q65i3shm@127.0.0.1:48342;transport=ws SIP/2.0
00009 1626865231 * ==> 127.0.0.1:33662 OPTIONS sip:h5bps1jp@127.0.0.1:33662;transport=ws SIP/2.0
00010 1626865231 * ==> 127.0.0.1:42702 OPTIONS sip:bsj8mjq7@127.0.0.1:42702;transport=ws SIP/2.0
00011 1626865232 * <== 127.0.0.1:48342 SIP/2.0 200 OK
00012 1626865232 * <== 127.0.0.1:33662 SIP/2.0 200 OK
00013 1626865262 * ==> .145:5060 OPTIONS sip:.145:5060 SIP/2.0
00014 1626865262 * <== .145:5060 SIP/2.0 200 OK
00015 1626865265 * ==> .145:5060 OPTIONS sip:.145:5060 SIP/2.0
00016 1626865266 * <== .145:5060 SIP/2.0 200 OK
00017 1626865272 * <== 127.0.0.1:58130 INVITE sip:91868086@ucpm.s**.com SIP/2.0
00018 1626865272 * ==> 127.0.0.1:58130 SIP/2.0 401 Unauthorized
00019 1626865272 * <== 127.0.0.1:58130 ACK sip:918680863226@ucpm.s.com SIP/2.0
00020 1626865272 * <== 127.0.0.1:58130 INVITE sip:918680863226@ucpdemoplatform.smblee.com SIP/2.0
00021 1626865272 * ==> 127.0.0.1:58130 SIP/2.0 100 Trying
00022 1626865273 * ==> .145:5060 INVITE sip:91868086@***.145:5060 SIP/2.0
00023 1626865273 * <== ***.145:5060 SIP/2.0 100 Trying
00024 1626865275 * <== ***.145:5060 SIP/2.0 183 Session Progress
00025 1626865275 * ==> 127.0.0.1:58130 SIP/2.0 183 Session Progress
00026 1626865275 * ==> 127.0.0.1:58130 SIP/2.0 183 Session Progress
00027 1626865281 * <== ***.145:5060 SIP/2.0 200 OK
00028 1626865281 * ==> .145:5060 ACK sip:.145:5070 SIP/2.0
00029 1626865281 * ==> 127.0.0.1:58130 SIP/2.0 200 OK
00030 1626865281 * <== 127.0.0.1:58130 ACK sip:127.0.0.1:5039;transport=ws SIP/2.0
00031 1626865282 * ==> 127.0.0.1:58130 OPTIONS sip:utq139nj@127.0.0.1:58130;transport=ws SIP/2.0
00032 1626865283 * <== 127.0.0.1:58130 SIP/2.0 200 OK
00033 1626865290 * ==> 127.0.0.1:59694 OPTIONS sip:jb98sgm8@127.0.0.1:59694;transport=ws SIP/2.0
00034 1626865290 * <== 127.0.0.1:59694 SIP/2.0 200 OK
00035 1626865291 * ==> 127.0.0.1:48342 OPTIONS sip:q65i3shm@127.0.0.1:48342;transport=ws SIP/2.0
00036 1626865291 * ==> 127.0.0.1:33662 OPTIONS sip:h5bps1jp@127.0.0.1:33662;transport=ws SIP/2.0
00037 1626865291 * ==> 127.0.0.1:42702 OPTIONS sip:bsj8mjq7@127.0.0.1:42702;transport=ws SIP/2.0
00038 1626865292 * <== 127.0.0.1:48342 SIP/2.0 200 OK
00039 1626865292 * <== 127.0.0.1:33662 SIP/2.0 200 OK
00040 1626865312 * <== 127.0.0.1:58130 BYE sip:127.0.0.1:5039;transport=ws SIP/2.0
00041 1626865312 * ==> 127.0.0.1:58130 SIP/2.0 200 OK
00042 1626865312 * ==> .145:5060 BYE sip:.146:5070 SIP/2.0
00043 1626865312 * <== ***.145:5060 SIP/2.0 200 OK
[Jul 21 11:05:13] == Manager ‘xivo_munin_user’ logged on from 127.0.0.1
Hello, sorry it doesn’t help to help you.
Have check the firewall rules for the RTP?
Audio is working with other network,patricularly few networks not receive rtp packets while using
wazo. the same network work with other webrtc solutions.
what would be a problem?
we are not able to identify it.
You need to check the SDP to find the issue with media in your SIP trame.
